The telecommunications industry keeps witnessing rapid changes in the way people and organizations communicate. Many of these changes spring from the explosive growth of the Internet and from applications based on the Internet Protocol (IP). The Internet has become an omnipresent means of communication, and the total amount of packet-based network traffic has quickly surpassed traditional voice (circuit-switched) network traffic, such as PSTN's (Public Switched Telephony Networks) and the like.
Technological advancements have helped telecommunication service providers, users, and suppliers realize that voice traffic and services may be one of the next major inroads to take full advantage of IP. This expectation is based on the impact of a new set of technologies generally referred to as Voice over IP (VOIP) or IP telephony.
VoIP supplies many unique capabilities to service providers and users who depend on IP or other packet-based networks. The most important benefits include the following: cost savings, open standards and multi-vendor interoperability, and integrated voice and data networks.
In the case of cost savings, users can reduce or eliminate toll charges associated with transporting calls over the PSTN by moving voice traffic to IP networks. In the case for open standards and multi-supplier interoperability, both service providers and/or users can purchase equipment from multiple suppliers and eliminate their dependency on proprietary solutions. Finally, in the case for integrated voice and data networks, service providers and/or users can build truly integrated networks for voice and data by making voice “just another IP application”. These integrated networks not only provide the quality and reliability of today's PSTN, but they also enable users to quickly and flexibly take advantage of new opportunities within the changing world of communications.
The first VoIP products were targeted at users looking to reduce telecommunication expenses by moving voice traffic to packet networks. To take advantage of favorable regulatory treatment of IP traffic and without any established standards, most early implementations were based on proprietary technology.
As in any emerging industry standard VoIP protocols were needed as packet telephony networks grew in numbers and size, and interconnection dependencies became more relevant. Eventually, the result was independent standards, each with its own unique characteristics. Network equipment suppliers and their users found out they needed to sort out the similarities and differences between different signaling and call-control protocols for VoIP, such as: H323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), and H248/Media Gateway Control (MEGACO), the Cisco SCCP Protocol, Skype's Protocol, and other proprietary protocols.
Thus, multiple VoIP protocols and architectures are presently deployed, and will continue to exist for the foreseeable future. Further, many networks will continue to be built using multiple VoIP protocols. As with current data networks which were built over time using multiple protocols and applications, the VoIP networks of today and tomorrow will continue to be constructed using the protocols and applications that best fit the associated technology and business requirements.
Because of the multiplicity of these VoIP protocols, users may have to investigate which VoIP protocols best serve their needs. The answer to this question may depend entirely on the unique requirements of each user's network implementation. VoIP comprises many standards and protocols. The following protocol definitions identify several examples of existing protocols (the protocols are listed in alphabetical order):                1. H248 is an International Telecommunication Union (ITU) Recommendation that defines “Gateway Control Protocol.” H248 is the result of a joint collaboration between the ITU and the Internet Engineering Task Force (IETF). It is also referred to as IETF RFC 2885 (MEGACO), which defines a centralized architecture for creating multimedia applications, including VoIP. In many ways, H248 builds on and extends MGCP.        2. H323 is an ITU Recommendation that defines “packet-based multimedia communications systems.” In other words, H323 defines a distributed architecture for creating multimedia applications, including VoIP.        3. MGCP, also known as IETF RFC 2705, defines a centralized architecture for creating multimedia applications, including VoIP.        4. SIP, also known as IETF RFC 2543 and/or 3261, defines a distributed architecture for creating multimedia applications, including VoIP.        
ITU is an international organization within the United Nations System where governments and the private sector coordinate global telecom networks and services. IETF refers to an Internet Engineering Task Force, which is a community of engineers that seeks to determine how the Internet and Internet protocols work, as well as to define the prominent standards. Also, it is important to note that Real Time Transport Protocol (RTP), also known as IETF RFC 1889, defines a transport protocol for real-time applications. Specifically, RTP provides the transport to carry the audio/media portion of VoIP communication. RTP is used by all VoIP signaling protocols.
One of the benefits of VoIP technology is that it allows networks to be built using either a centralized or a distributed architecture. In general, centralized architectures are associated with H248 and MGCP. These protocols were designed for a centralized device—called a media gateway controller or call agent—that handles switching logic and call control. The centralized device talks to media gateways, which route and transmit the audio/media portion of the calls (the actual voice information).
In centralized architectures, the network intelligence is centralized and endpoints are relatively dumb (with limited or no native features). Although most centralized VoIP architectures may be using H248 and MGCP protocol, it may also be possible to build H323 or SIP networks in a centralized fashion using Back-to-Back User Agents (B2BUAs) or Gate Keeper Routed Call Signaling (GKRCS), respectively.
Distributed architectures are associated with H323 and SIP protocols. These protocols allow network intelligence to be distributed between endpoints and call-control devices. Intelligence in this instance refers to call state, calling features, call routing, provisioning, billing, or any other aspect of call handling. The endpoints can be VoIP gateways, IP phones, media servers, or any device that can initiate and terminate a VoIP call. The call-control devices are called gatekeepers in an H323 network, and proxy or redirect servers in a SIP network.
H323 was originally created to provide a mechanism for transporting multimedia applications over local-area networks (LANs). Although H323 is still used by numerous vendors for videoconferencing applications, it has rapidly evolved to address the growing needs of VoIP networks. Because of its early availability and these advancements, H323 is currently the most widely used VoIP signaling and call-control protocol, with international and domestic carriers relying on it to handle billions of minutes of use each year. H323 defines Registration, Admission, and Status (RAS) protocol for call routing. H323 is based on the Integrated Services Digital Network (ISDN) Q.931 protocol, which allows it to easily interoperate with legacy voice networks such as the PSTN or Signaling System 7 (SS7).
H248 and MGCP were designed to provide an architecture where call control and services could be centrally added to a VoIP network. In that sense, an architecture using these protocols closely resembles the existing PSTN architecture and services. H248 and MGCP define most aspects of signaling using a package model. These packages define commonly used functionality, such as PSTN signaling, line-side device connectivity, and features such as transfer and hold. In addition, session definition protocol (SDP) is used to convey capabilities exchange.
On the other hand, SIP was designed as a multimedia protocol that could take advantage of the architecture and messages found in popular Internet applications. By using a distributed architecture—with universal resource locators (URLs) for naming and text-based messaging—SIP attempts to take advantage of the Internet model for building VoIP networks and applications. In addition to VoIP, SIP may be used for video-conferencing and instant messaging. As a protocol, SIP defines how sessions are to be set up and torn down. SIP may be used to invite a user to take part in a point-to-point or a uni-cast session. It utilizes other IETF protocols to define other aspects of VoIP and multimedia sessions, such as SDP for capabilities exchange, URLs for addressing, Domain Name Systems (DNS's) for service location, and Telephony Routing Over IP (TRIP) for call routing.
VoIP networks are expected to continue to be made up of multiple protocols, because suppliers support for each protocol differs and users have varying business requirements. Having multiple protocols gives users the flexibility they need to connect for services from multiple providers or they may be faced with choices about how to interconnect segments using differing VoIP protocols. For example, these choices may be as follows: Translation through Time Division Multiplexing (TDM), Single Protocol Architecture, or Protocol Translation.
The TDM choice allows a provider/user to translate from one protocol domain to another. But this translation typically occurs in two steps as follows: VoIP no. 1⇄TDM⇄VoIP no. 2. Hence a delay is introduced, and this is usually considered only as a short-term solution.
The Single Protocol Architecture choice moves all of a provider/user VoIP devices and services to a single protocol, thus simplifying the network as a whole. A downside to this choice is that it might not be possible to migrate existing equipment to support the new protocol, a situation that can limit the provider/user's ability to take advantage of some existing services. In addition, it may limit the potential connectivity to other networks that are using other VoIP signaling protocols.
The Protocol Translation choice allows a provider/user to use IP-based protocol translators to interconnect two or more VoIP protocol domains. IP translators allow a provider/user to retain the flexibility of using multiple VoIP protocols, do not introduce the delay problems that additional TDM interconnections do, and do not require a wholesale replacement or swap of existing equipment. A downside to this choice is that there is no standard for protocol translation, so not all VoIP protocol translators are exactly the same. Although the IETF has attempted to define a model for translating H323 to SIP, such translation schemes are complex, involving much more than just building a protocol-translation box.
All the above choices seem to bring solutions with additional complexity to a provider/user's already complex situation while attempting to streamline the multiple VoIP protocol reality within its network. Therefore, there is a need for a solution that would accommodate all the multiple VoIP protocols while minimizing the addition of complexity, for example, without introducing communication delays, without migrating all existing equipment to a new protocol, without imposing a new standard for a protocol translator, or without requiring that a VoIP phone device resides in a centralized or in a decentralized architecture. Such a solution would, for example, retain the flexibility of using all multiple VoIP protocols, avoid limiting the potential connectivity to other networks, etc. In addition, it would be desirable in some embodiments to introduce VoIP phone devices that actually incorporate multiple competing protocols in VoIP, such as H248, H. 323, MGCP, SIP, and the like.